📄 ss7.conf.template.single-link
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[linkset-siuc]; The linkset is enabledenabled => yes; The end-of-pulsing (ST) is not used to determine when incoming address is completeenable_st => no; Reply incoming call with CON rather than ACM and ANMuse_connect => yes; The CIC hunting policy (even_mru, odd_lru, seq_lth, seq_htl) is even CIC numbers, most recently usedhunting_policy => even_mru; Incoming calls are placed in the ss7 context in the asterisk dialplancontext => ss7; The language for this context is dalanguage => da; The value and action for t35. Value is in msec, action is either st or timeout; If you use overlapped dialling dial plan, you might choose: t35 => 4000,stt35 => 15000,timeout; The subservice field: national (8), international (0), auto or decimal/hex value; The auto means that the subservice is obtained from first received SLTMsubservice => auto; The host running the mtp3 service; mtp3server => localhost[link-l1]; This link belongs to linkset siuclinkset => siuc; The speech/audio circuit channels on this linkchannels => 1-15,17-31; The signalling channelschannel => 16; To use the remote mtp3 service, use 'schannel => remote,16'; The first CICfirstcic => 1; The link is enabledenabled => yes; Echo cancellation; echocancel can be one of: no, 31speech (enable only when transmission medium is 3.1Khz speech), allwaysechocancel => no; echocan_train specifies training period, between 10 to 100 msecechocan_train => 350; echocan_taps specifies number of taps, 32, 64, 128 or 256echocan_taps => 128[host-gentoo1]; chan_ss7 auto-configures by matching the machines host name with the host-<name>; section in the configuration file, in this case 'gentoo1'. The same; configuration file can thus be used on several hosts. ; The host is enabledenabled => yes; The point code for this SS7 signalling point is 0x8e0opc => 0x8e0; The destination point (peer) code is 0x3fff for linkset siucdpc => siuc:0x3fff; If the link connected to an STP (with point code 0x123), you may use; dpc => siuc:0x3ff,l1:0x123; Syntax: links => link-name:digium-connector-no; The links on the host is 'l1', connected to span/connector #1links => l1:1; The SCCP global title: translation-type, nature-of-address, numbering-plan, addressglobaltitle => 0x00, 0x04, 0x01, 4546931411ssn => 7[jitter];------------------------------ JITTER BUFFER CONFIGURATION --------------------------; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled.; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to "no".; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000.; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed.; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".;-----------------------------------------------------------------------------------
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